similar to: Garbled voice on long distance calls

Hi, We have Asterisk, running on a P4 box running Suse 9.1, making calls using IAX through SimpleTelecom and Nufone. What we are looking for is toll quality voice. The problem is that voice over calls routed through SimpleTelecom and nNufone occassionally breaks. We also have a digium card and the calls over the digium card using the Zaptel Interface have a very good quality. We

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Thanks, Jim

Folks, I've ordered a new account from Nufone last month. Transferred money to Nufone through their paypal account. I had communication with Nufone sales up until two weeks back. Since then there were no replies to my emails. I am afraid with this kind of unresponsiveness how one would run a reliable service with this company. Have no bad feeling with Jeremy as the author of widely used h323

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

I just upgraded to RC1 from a two-three month old CVS , and noticed that during IAX2 calls to my service provider there are periods (usually less than 10 seconds long, minutes apart) during which the caller can not hear me, but I can hear the caller fine. Inter-office calls (SIP-to-SIP) does not appear to have this issue. Has any other users experienced this? Marcus Adolfsson TreoCentral

problem - occasional garbled calls, mostly remote users. T1 connected to PSTN, SIP over local LAN and internet to "remote users". NAT at local firewall and at remotes. There is no traffic shaping in place, no QoS. Most are Polycom phones, two Aastra's. Start with QoS on LAN switches? No 2x4's please, start with 1x4's. joe a.

Hi All; Can I run the IAX on TCP port instead of UDP port? If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? Same thing

Not to point the finger at Teliax, but I'm having some unique problems with their service that are as yet unexplained. Incoming calls are fine. Outgoing calls don't work, though they did at one time. As of today, I'm running the latest code from CVS. -- Called teliax/13143212222 -- Call accepted by 208.139.204.245 <http://208.139.204.245> (format ulaw) -- Format for call is

Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf

Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond....) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line

On 2018-08-27, Stuart Henderson <stu at spacehopper.org> wrote: > On 2018-08-27, Zach Cheung <kuroro.zhang at gmail.com> wrote: >> After upgrading my VMware Fusion (10.1.3) Arch Guest to the latest with >> OpenSSH upgraded from 7.7p1 to 7.8p1, found that ssh from macOS Sierra >> (10.12.6) host to Arch guest via local NAT port forwarding failed, but via >>

Hi All, I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used for PSTN calls via IAX2. Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). We've no E1/T1 links, everything is IP based. My boss complains that many of the calls he holds with others has a bad quality. He also says that its not just him. My iax.conf file

I have a DID with VoicePulse Connect, but the sound quality is horrible, it is often choppy and the caller's voice cuts out for 2-3 seconds at least once a minute, I have contacted VoicePulse many times, and they do not do anything about it! Does anyone have any similar problems? It isnt my Asterisk config because I have 0 problems using NuFone.

After upgrading my VMware Fusion (10.1.3) Arch Guest to the latest with OpenSSH upgraded from 7.7p1 to 7.8p1, found that ssh from macOS Sierra (10.12.6) host to Arch guest via local NAT port forwarding failed, but via Arch LAN IP worked, downgraded OpenSSH from 7.8p1 to 7.7p1 fixed the problem. Any idea about this bug? VMware Fusion NAT Port Forwarding setup: VMware Fusion ? NAT Port Forwarding

Grasping at straws here...is anyone using a Dell PowerConnect 2224 24-port unmanaged 10/100 switch in a deployment? I have two separate asterisk installations with bad one-way audio where the only common elements left are the Dell switches and Polycom IP-500 phones. Two different ITSPs - one location uses IAX2 to the outside world, the other SIP. Two different bandwidth providers, one SDSL 1.5

On Mon, 27 Aug 2018, Stuart Henderson wrote: > On 2018-08-27, Zach Cheung <kuroro.zhang at gmail.com> wrote: > > After upgrading my VMware Fusion (10.1.3) Arch Guest to the latest with > > OpenSSH upgraded from 7.7p1 to 7.8p1, found that ssh from macOS Sierra > > (10.12.6) host to Arch guest via local NAT port forwarding failed, but via > > Arch LAN IP worked,

Can anyone point me to some sample Cisco QoS configurations suitable for IAX2? I've looked through Cisco's site, and get overwhelmed with the level of documentation (too much of a good thing). My PSTN gateway and PBX (both *) are connected via 2xT1 (per-packet load balancing) between a Cisco 7206 and a 3640. When the total bandwidth pushes much past 50%, I start getting some crazy

Greetings... ** Short Description / Abstract ** Using my _internal_ WiFi card, OpenSSH succeeds to local (internal) LAN hosts, but hangs after authentication to external LAN hosts; however PuTTY works for all hosts. Using an _external_ WiFi card, OpenSSH does succeed to all LAN hosts (as did PuTTY, still, as well). ** Longer Description and Details ** *** HW Description *** I have a laptop